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1 /* | |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | |
3 * | |
4 * Use of this source code is governed by a BSD-style license | |
5 * that can be found in the LICENSE file in the root of the source | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_ | |
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_ | |
13 | |
14 #include <memory> | |
15 | |
16 #include "webrtc/common_audio/channel_buffer.h" | |
17 #include "webrtc/modules/audio_processing/utility/audio_ring_buffer.h" | |
18 | |
19 namespace webrtc { | |
20 | |
21 // The callback function to process audio in the time domain. Input has already | |
22 // been windowed, and output will be windowed. The number of input channels | |
23 // must be >= the number of output channels. | |
24 class BlockerCallback { | |
25 public: | |
26 virtual ~BlockerCallback() {} | |
27 | |
28 virtual void ProcessBlock(const float* const* input, | |
29 size_t num_frames, | |
30 size_t num_input_channels, | |
31 size_t num_output_channels, | |
32 float* const* output) = 0; | |
33 }; | |
34 | |
35 // The main purpose of Blocker is to abstract away the fact that often we | |
36 // receive a different number of audio frames than our transform takes. For | |
37 // example, most FFTs work best when the fft-size is a power of 2, but suppose | |
38 // we receive 20ms of audio at a sample rate of 48000. That comes to 960 frames | |
39 // of audio, which is not a power of 2. Blocker allows us to specify the | |
40 // transform and all other necessary processing via the Process() callback | |
41 // function without any constraints on the transform-size | |
42 // (read: |block_size_|) or received-audio-size (read: |chunk_size_|). | |
43 // We handle this for the multichannel audio case, allowing for different | |
44 // numbers of input and output channels (for example, beamforming takes 2 or | |
45 // more input channels and returns 1 output channel). Audio signals are | |
46 // represented as deinterleaved floats in the range [-1, 1]. | |
47 // | |
48 // Blocker is responsible for: | |
49 // - blocking audio while handling potential discontinuities on the edges | |
50 // of chunks | |
51 // - windowing blocks before sending them to Process() | |
52 // - windowing processed blocks, and overlap-adding them together before | |
53 // sending back a processed chunk | |
54 // | |
55 // To use blocker: | |
56 // 1. Impelment a BlockerCallback object |bc|. | |
57 // 2. Instantiate a Blocker object |b|, passing in |bc|. | |
58 // 3. As you receive audio, call b.ProcessChunk() to get processed audio. | |
59 // | |
60 // A small amount of delay is added to the first received chunk to deal with | |
61 // the difference in chunk/block sizes. This delay is <= chunk_size. | |
62 // | |
63 // Ownership of window is retained by the caller. That is, Blocker makes a | |
64 // copy of window and does not attempt to delete it. | |
65 class Blocker { | |
66 public: | |
67 Blocker(size_t chunk_size, | |
68 size_t block_size, | |
69 size_t num_input_channels, | |
70 size_t num_output_channels, | |
71 const float* window, | |
72 size_t shift_amount, | |
73 BlockerCallback* callback); | |
74 | |
75 void ProcessChunk(const float* const* input, | |
76 size_t chunk_size, | |
77 size_t num_input_channels, | |
78 size_t num_output_channels, | |
79 float* const* output); | |
80 | |
81 private: | |
82 const size_t chunk_size_; | |
83 const size_t block_size_; | |
84 const size_t num_input_channels_; | |
85 const size_t num_output_channels_; | |
86 | |
87 // The number of frames of delay to add at the beginning of the first chunk. | |
88 const size_t initial_delay_; | |
89 | |
90 // The frame index into the input buffer where the first block should be read | |
91 // from. This is necessary because shift_amount_ is not necessarily a | |
92 // multiple of chunk_size_, so blocks won't line up at the start of the | |
93 // buffer. | |
94 size_t frame_offset_; | |
95 | |
96 // Since blocks nearly always overlap, there are certain blocks that require | |
97 // frames from the end of one chunk and the beginning of the next chunk. The | |
98 // input and output buffers are responsible for saving those frames between | |
99 // calls to ProcessChunk(). | |
100 // | |
101 // Both contain |initial delay| + |chunk_size| frames. The input is a fairly | |
102 // standard FIFO, but due to the overlap-add it's harder to use an | |
103 // AudioRingBuffer for the output. | |
104 AudioRingBuffer input_buffer_; | |
105 ChannelBuffer<float> output_buffer_; | |
106 | |
107 // Space for the input block (can't wrap because of windowing). | |
108 ChannelBuffer<float> input_block_; | |
109 | |
110 // Space for the output block (can't wrap because of overlap/add). | |
111 ChannelBuffer<float> output_block_; | |
112 | |
113 std::unique_ptr<float[]> window_; | |
114 | |
115 // The amount of frames between the start of contiguous blocks. For example, | |
116 // |shift_amount_| = |block_size_| / 2 for a Hann window. | |
117 size_t shift_amount_; | |
118 | |
119 BlockerCallback* callback_; | |
120 }; | |
121 | |
122 } // namespace webrtc | |
123 | |
124 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_BLOCKER_H_ | |
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